Current wireless communication systems provide the ability for users to communicate to and from wireless or mobile users. There are generally two types of wireless communication systems, circuit-switched (“CS”) and packet-switched (“PS”).
In typical circuit-switched wireless communication systems, the mobile switching center (“MSC”) connects the landline public switched telephone network (“PSTN”) system to the wireless communication system. The mobile switching center is typically split into an mobile switching center server and a media gateway (“MGW”), and incorporates the bearer independent call control (“BICC”) or the integrated services digital network user part (“ISUP”) call control protocol for call delivery between mobile switching centers.
The current approach to introducing internet protocol (“IP”) multimedia services for universal mobile telecommunications service (“UMTS”) and code division multiple access (“CDMA”) third generation (“3G”) systems is to define a brand new internet protocol multimedia subsystem (“IMS”), comprised of a set of internet protocol connected network entities within the internet protocol multimedia subsystem using packet-switched services. These network entities provide internet protocol multimedia features and services using the session initiation protocol (“SIP”) as the primary vehicle for call control.
As the network entities become more centralized and employ disparate codecs to communicate, the difficulty involved in allowing end users to communicate increases. It is desirable to enable the negotiation of codecs between remote network resources and end users to optimally configure a conference call.
Wireless service operators are looking for efficient solutions to: 1) utilize the packet transport for circuit voices in the backbone core network; and 2) be able to negotiate and modify codecs to facilitate both Transcoder Free Operation (TrFO) and Remote Transcoder Operation (RTO) over packet networks. The TrFO and RTO bring the benefits of voice quality improvement, saving of transport facility and reduction of network resources.
In the existing wireless circuit network, conferencing operation is a value-added service. While evolving the TDM based transport to packet-based network, it is highly desirable to find a solution that supports fast setup of a conference bridge that models an existing circuit conference operation, minimizes usage of network resource and maintains good voice quality. The solution should also support SIP enabled packet mobiles with end-to-end VoIP applications.
Thus there is a need in the prior art for fast network SIP/SDP (Session Initiation Protocol/Session Description Protocol) procedures for conference operations upon request from end user with optimization of network resources.